Replace scoped_ptr with unique_ptr everywhere

webrtc/test/call_test.h

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
@@ skipping 64 matching lines @@
 
   void CreateVideoStreams();
   void CreateAudioStreams();
   void Start();
   void Stop();
   void DestroyStreams();
   void SetFakeVideoCaptureRotation(VideoRotation rotation);
 
   Clock* const clock_;
 
-  rtc::scoped_ptr<Call> sender_call_;
-  rtc::scoped_ptr<PacketTransport> send_transport_;
+  std::unique_ptr<Call> sender_call_;
+  std::unique_ptr<PacketTransport> send_transport_;
   VideoSendStream::Config video_send_config_;
   VideoEncoderConfig video_encoder_config_;
   VideoSendStream* video_send_stream_;
   AudioSendStream::Config audio_send_config_;
   AudioSendStream* audio_send_stream_;
 
-  rtc::scoped_ptr<Call> receiver_call_;
-  rtc::scoped_ptr<PacketTransport> receive_transport_;
+  std::unique_ptr<Call> receiver_call_;
+  std::unique_ptr<PacketTransport> receive_transport_;
   std::vector<VideoReceiveStream::Config> video_receive_configs_;
   std::vector<VideoReceiveStream*> video_receive_streams_;
   std::vector<AudioReceiveStream::Config> audio_receive_configs_;
   std::vector<AudioReceiveStream*> audio_receive_streams_;
 
-  rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
+  std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
   test::FakeEncoder fake_encoder_;
   std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_;
   size_t num_video_streams_;
   size_t num_audio_streams_;
 
  private:
   // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.
   // These methods are used to set up legacy voice engines and channels which is
   // necessary while voice engine is being refactored to the new stream API.
   struct VoiceEngineState {
     VoiceEngineState()
         : voice_engine(nullptr),
           base(nullptr),
           codec(nullptr),
           channel_id(-1) {}
 
     VoiceEngine* voice_engine;
     VoEBase* base;
     VoECodec* codec;
     int channel_id;
   };
 
   void CreateVoiceEngines();
   void DestroyVoiceEngines();
 
   VoiceEngineState voe_send_;
   VoiceEngineState voe_recv_;
 
   // The audio devices must outlive the voice engines.
-  rtc::scoped_ptr<test::FakeAudioDevice> fake_send_audio_device_;
-  rtc::scoped_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
+  std::unique_ptr<test::FakeAudioDevice> fake_send_audio_device_;
+  std::unique_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
 };
 
 class BaseTest : public RtpRtcpObserver {
  public:
   explicit BaseTest(unsigned int timeout_ms);
   virtual ~BaseTest();
 
   virtual void PerformTest() = 0;
   virtual bool ShouldCreateReceivers() const = 0;
 
@@ skipping 37 matching lines @@
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_